1. Field of the Invention
The present invention relates to a redundancy gateway system that includes plural gateway units configured in a multiplex manner for plural systems and that carries out data communications such as audio data communications and video data communication between a packet network and a TDM network such as a telephone network.
2. Description of the Related Art
Generally, a gateway unit that connects a packet network and a telephone network with each other and relays audio data has a function of encoding an audio signal received from the telephone network by an encoder, then packetizing the audio data in accordance with RTP (Real-time Transport Protocol) and sending the packetized data to the packet network. Conversely, the gateway unit also has a function of decoding an audio packet received from the packet network by a decoder and then sending the decoded audio data to the telephone network. To restrain the influence of failure in the gateway unit on user's communication, the gateway unit is often has a multiple redundant configuration of duplex or more. For example, in a duplex redundant configuration in which one of two gateway units is set for an operation system and the other one is set for a standby system, if failure occurs in the operation system or if maintenance of the unit is necessary, system switching is carried out between the units in the operation system and the standby system. A method has been proposed to prevent interruption of communication in this system switching.
In the method disclosed in Japanese Patent Kokai No. 2005-57461, when switching between the operation system and the standby system is carried out, RTP/RTCP session information except for information related to the time stamp and sequence number, and connectivity information are transferred from the operation system to the standby system. Thus, the gateway unit switched to the operation system can start a decoding operation using a parameter value set for an RTP packet sent from an IP terminal as an initial value, and the operation system and the standby system can be switched without interrupting the audio.
However, the conventional technique cannot achieve the elimination of short interruption of speaking. In the disclosed method, since the time stamp value and sequence number of the RTP session information to be sent to the packet network are not taken over, occurrence of short interruption of speaking due to detection of discontinuity in the time stamp value and sequence number of the RTP packet in the receiving device cannot be avoided. Also, with respect to processing to perform audio processing of data received from the packet network and then output the audio data to the telephone network, synchronization of processing cannot be taken between the operation system and the standby system. Therefore, short interruption or duplication of speaking occurs in system switching. Moreover, since the new operation system unit starts operating with an initial state irrespective of the state of the former operation system unit, degradation of the speaking quality cannot be avoided until the new operation system restores a state equivalent to the state of the former operation system. For example, speaking may be interrupted while packets remaining in the jitter buffer of the former operation system are discarded, or a time of approximately 500 to 800 ms may be necessary in order to reach a state where echo can be properly eliminated as optimization control is reset in the echo canceller of the new operation system, and the speech during this period may not sound normal.